mirror of https://git.wownero.com/dsc/mxe.git
113 lines
4.3 KiB
Diff
113 lines
4.3 KiB
Diff
This file is part of MXE.
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See index.html for further information.
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Taken from libdca svn: svn://svn.videolan.org/libdca/trunk.
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r84 | gbazin | 2008-06-01 16:13:33 +0000 (Sun, 01 Jun 2008) | 3 lines
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* libdca/parse.c: Change output normalisation factor from 3/2 to sqrt(2).
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Thanks to Alexander E. Patrakov for finding that this is the proper normalisation factor.
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Fixed a bug where the output bias wasn't applied when downmixing wasn't being done.
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--- libdca.orig/libdca/parse.c
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+++ libdca/libdca/parse.c
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@@ -59,12 +59,11 @@
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static int decode_blockcode (int code, int levels, int *values);
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static void qmf_32_subbands (dca_state_t * state, int chans,
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- double samples_in[32][8], sample_t *samples_out,
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- double rScale, sample_t bias);
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+ double samples_in[32][8], sample_t *samples_out);
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static void lfe_interpolation_fir (int nDecimationSelect, int nNumDeciSample,
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double *samples_in, sample_t *samples_out,
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- double rScale, sample_t bias );
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+ sample_t bias);
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static void pre_calc_cosmod( dca_state_t * state );
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@@ -123,7 +122,9 @@
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bitstream_get (state, 1);
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*frame_length = (bitstream_get (state, 7) + 1) * 32;
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+ if (*frame_length < 6 * 32) return 0;
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frame_size = bitstream_get (state, 14) + 1;
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+ if (frame_size < 96) return 0;
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if (!state->word_mode) frame_size = frame_size * 8 / 14 * 2;
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/* Audio channel arrangement */
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@@ -981,14 +982,7 @@
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/* 32 subbands QMF */
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for (k = 0; k < state->prim_channels; k++)
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{
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- /*static double pcm_to_float[8] =
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- {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
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-
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- qmf_32_subbands (state, k,
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- subband_samples[k],
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- &state->samples[256*k],
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- /*WTF ???*/ 32768.0*3/2/*pcm_to_float[state->source_pcm_res]*/,
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- 0/*state->bias*/);
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+ qmf_32_subbands (state, k, subband_samples[k], &state->samples[256*k]);
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}
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/* Down/Up mixing */
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@@ -1000,6 +994,10 @@
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{
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dca_downmix (state->samples, state->amode, state->output, state->bias,
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state->clev, state->slev);
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+ } else if (state->bias)
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+ {
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+ for ( k = 0; k < 256*state->prim_channels; k++ )
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+ state->samples[k] += state->bias;
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}
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/* Generate LFE samples for this subsubframe FIXME!!! */
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@@ -1011,8 +1009,7 @@
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lfe_interpolation_fir (state->lfe, 2 * state->lfe,
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state->lfe_data + lfe_samples +
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2 * state->lfe * subsubframe,
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- &state->samples[256*i_channels],
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- 8388608.0, state->bias);
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+ &state->samples[256*i_channels], state->bias);
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/* Outputs 20bits pcm samples */
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}
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@@ -1142,9 +1139,9 @@
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}
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static void qmf_32_subbands (dca_state_t * state, int chans,
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- double samples_in[32][8], sample_t *samples_out,
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- double scale, sample_t bias)
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+ double samples_in[32][8], sample_t *samples_out)
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{
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+ static const double scale = 1.4142135623730951 /* sqrt(2) */ * 32768.0;
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const double *prCoeff;
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int i, j, k;
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double raXin[32];
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@@ -1211,7 +1208,7 @@
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/* Create 32 PCM output samples */
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for (i=0;i<32;i++)
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- samples_out[nChIndex++] = subband_fir_hist2[i] / scale + bias;
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+ samples_out[nChIndex++] = subband_fir_hist2[i] / scale;
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/* Update working arrays */
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for (i=511;i>=32;i--)
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@@ -1225,7 +1222,7 @@
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static void lfe_interpolation_fir (int nDecimationSelect, int nNumDeciSample,
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double *samples_in, sample_t *samples_out,
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- double scale, sample_t bias)
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+ sample_t bias)
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{
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/* samples_in: An array holding decimated samples.
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* Samples in current subframe starts from samples_in[0],
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@@ -1235,6 +1232,7 @@
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* samples_out: An array holding interpolated samples
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*/
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+ static const double scale = 8388608.0;
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int nDeciFactor, k, J;
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const double *prCoeff;
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